Check X-lite sending register request or not to asterisk buy checking
the asterisk console,if not there would some problem in X-lite
configuration settings,if sending check the console and see what error
logs you are getting..
Thanks
Nikhil
On 11/18/2010 04:06 PM, Phuong Hoang wrote:
Hi
Hello,
We tried to upgrade our Asterisk from 1.6.2.13 to 1.6.2.14, after trying
to install iksemel (jabber support) and spandsp, but now Asterisk
doesn't work anymore and we can't get it to run, althorugh we tried to
remove it completely and reinstall 1.6.2.13.
when trying to start it via
Hello Asterisk community,
We are having some problems with crashes in Asterisk, my asterisk
versions are 1.4.24.1 and 1.4.23.2. I have found this:
~/work/asterisk-branch-1.4$ svn log -c 260345
r260345 | mmichelson |
Do your asterisk logs say anything -- /var/log/asterisk/messages or
full? Also, what happens if you do asterisk -c this may help you
figure things out.
Michael voip.quest...@gmail.com wrote:
Hello,
We tried to upgrade our Asterisk from 1.6.2.13 to 1.6.2.14, after trying
to install
- Original Message -
Hello,
We tried to upgrade our Asterisk from 1.6.2.13 to 1.6.2.14, after
trying
to install iksemel (jabber support) and spandsp, but now Asterisk
doesn't work anymore and we can't get it to run, althorugh we tried to
remove it completely and reinstall
It's a peer name defined below in sip.conf. You may skip secret if it is
specified in peer section. I don't know of any other meanings.
For example,
register = mypeer?u...@host
[mypeer]
type=peer
defaultuser=user
secret=blah
...
This syntax exists since 1.6.2.
21.10.2010 17:31, Guillaume Bour
Also, what happens if you do asterisk -c this may help you
figure things out.
Hi,
These are the WARNINGSI found in /var/log/asterisk/messages after
running the above command:
[Nov 22 12:10:19] WARNING[2316] udptl.c: T38FaxUdpEC in udptl.conf is no
longer supported; use the t38pt_udptl
- Original Message -
Hi all!
A few days I have problems connecting to the Internet on my house and
since then my local SIP extensions are no longer registered against
the
local Asterisk server.
I'm using Asterisk 1.4.24.1. I was researching on the Internet and I
found that it
Hi, Phil.
A few days I have problems connecting to the Internet on my house
and since then my local SIP extensions are no longer registered
against the local Asterisk server.
I'm using Asterisk 1.4.24.1. I was researching on the Internet and I
found that it can be related to a bug of
Hi Moises,
Thanks for your opinion.
However I still wouldn't want to agree that reducing debug logging is a
solution. Let me explain why, we are driving Asterisk using AMI and verbose
logging is simply not enough to investigate issues that arises with our
software or Asterisk itself. Also we are
- Original Message -
Hi, Phil.
A few days I have problems connecting to the Internet on my house
and since then my local SIP extensions are no longer registered
against the local Asterisk server.
I'm using Asterisk 1.4.24.1. I was researching on the Internet and
I
Hasn't this been fixed in later versions?
1.4.37 is current, or at least it was in the last few days.
Upgrading with no reason isn't suggested, but in this case you have a
good reason, and if you dig deep enough you may find the fix is already
in place.
John Novack
Danny Dias wrote:
Hello
Hi All,
We have a requirement to record over 60 simultaneous calls. Our recording
facilities are implemented using Monitor() over AMI. The thing we have
noticed that making 60 simultaneous call recordings using wav CPU load is
significantly higher (around 2 times more) than using gsm. Even
Hello John,
What i am asking is if i can apply this patch manually or something like
this without making any upgrade of Asterisk, has anyone done this before?
Or i have to upgrade my Asterisk versions...i don't really want to do
this...
Thanks in Advance!
2010/11/22 John Novack
Does you Asterisk server point to an internal DNS or to your
router ?
The /etc/resolv.conf of the host on which I installed Asterisk
points to an internal DNS. Is there a parameter in the Asterisk
configuration where also I have to force the use of an internal
DNS server?
Do your
What format are the actual calls in? Are they in G.711u/a format or
are they in something else (perhaps gsm?) format? I'm asking to find
out if Asterisk would need to transcode them.
On Mon, Nov 22, 2010 at 6:47 AM, Vilius Adamkavicius
vilius.adamkavic...@invade.net wrote:
Hi All,
We have a
Danny Dias wrote:
Hello John,
What i am asking is if i can apply this patch manually or something
like this without making any upgrade of Asterisk, has anyone done this
before?
I can't answer that question.
Or i have to upgrade my Asterisk versions...i don't really want to do
this...
2010/11/22 John Novack jnov...@stromberg-carlson.org
Danny Dias wrote:
Hello John,
What i am asking is if i can apply this patch manually or something like
this without making any upgrade of Asterisk, has anyone done this before?
I can't answer that question.
ummm why not? is
Hi Michael,
With regards the following error:
'res_config_odbc.so': /usr/lib/asterisk/modules/res_config_odbc.so: undefined
symbol: ast_odbc_clear_cache
You can fix that one by modifying /etc/asterisk/modules.conf and uncommenting
the following 2 lines:
preload = res_odbc.so
preload =
Hi Joel,
We have a meetme on which we are landing two G.711 alaw calls, one coming
from TDM another from SIP. Once we those parties are in the conference we
are adding one more leg using Local channel and starting to record it.
Surely it would be logical if it would be less overhead recording
Application of a patch to any one-or-more-off version of asterisk can be a
Russian roulette proposition; If you're applying 1-off you're pretty
safe. The more versions between the patch and where you are, the more
bullets you are loading into the gun.
The best (IMO) procedure for this or any
On Mon, Nov 22, 2010 at 9:50 AM, Danny Dias ing.diasda...@gmail.com wrote:
2010/11/22 John Novack jnov...@stromberg-carlson.org
Danny Dias wrote:
Hello John,
What i am asking is if i can apply this patch manually or something like
this without making any upgrade of Asterisk, has anyone
On Mon, Nov 22, 2010 at 8:47 AM, Vilius Adamkavicius
vilius.adamkavic...@invade.net wrote:
Hi All,
We have a requirement to record over 60 simultaneous calls. Our recording
facilities are implemented using Monitor() over AMI. The thing we have
noticed that making 60 simultaneous call
On Sun, Nov 21, 2010 at 8:14 PM, Daniel Bareiro daniel-lis...@gmx.net wrote:
Hi all!
A few days I have problems connecting to the Internet on my house and
since then my local SIP extensions are no longer registered against the
local Asterisk server.
You have to be a bit more specific. For
Hi David,
Looking at MOS G.711alaw wav most definitely has the higher score than gsm.
Moreover recording in gsm is more CPU intense than wav. Therefore your
suggestion to do more CPU intense recording and afterwards use system
resources to convert it back to wav is not a solution. Also some of
Hi Michael,
With regards the following error:
'res_config_odbc.so': /usr/lib/asterisk/modules/res_config_odbc.so: undefined
symbol: ast_odbc_clear_cache
You can fix that one by modifying /etc/asterisk/modules.conf and uncommenting
the following 2 lines:
preload = res_odbc.so
preload
Someone has hacked into our system and is making calls overseas.
How can I:
1. Find out the where the calls are originating from?
2. Block all calls that are not authorized?
Our system is in the USA.
Only calls from inside our LAN are allowed.
Thank you,
Gary Kuznitz
--
I've had phones before where, with the phone on-hook, it still implements
the local dialplan. E.g., if I dialed 0 (on-hook), after three seconds,
it would dial the operator, and have the call on speakerphone. Does
Polycom allow this functionality? Clearly, not a necessary feature... but
it
On Mon, Nov 22, 2010 at 03:28:27PM +, Vilius Adamkavicius wrote:
Hi David,
Looking at MOS G.711alaw wav most definitely has the higher score than gsm.
Moreover recording in gsm is more CPU intense than wav. Therefore your
suggestion to do more CPU intense recording and afterwards use
WAV or wav? One of these has GSM-encoding inside a WAV formatted
envelope. That said, I wouldn't expect that to have any noticeable
CPU utilization above that of GSM. If you are using the non-GSM
version of WAV, then I am as baffled as you - hopefully someone who
knows more about this can help.
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gary Kuznitz
Sent: Monday, November 22, 2010 10:23 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Someone has hacked into our system
Hi, Alejandro.
A few days I have problems connecting to the Internet on my house
and since then my local SIP extensions are no longer registered
against the local Asterisk server.
You have to be a bit more specific. For example is your Asterisk box
behind a router/nat? Or does your
Blocking udp 5060 in the packet filter in unwanted directions should
keep asterisk from setting up SIP connections.
The real remedy is to figure out how the hacker got in and close the
backdoor.
I think a lot of us would be interested in what was the vulnerability.
And if it turns out that it
On Mon, Nov 22, 2010 at 11:44 AM, Daniel Bareiro daniel-lis...@gmx.net wrote:
Hi, Alejandro.
A few days I have problems connecting to the Internet on my house
[...]
It would appear that the server for some reason was 'locked'. For
example, when I try to register from Twinkle softphone, I get
We are using wav, not WAV. I believe WAV is the one with GSM. Its a very
good idea to compare WAV against wav, will run some tests and come back with
outcome, will try Tzafrir's suggestion as well.
Thanks guys
Vilius.
On 22 November 2010 16:31, Joel Maslak jmas...@antelope.net wrote:
WAV or
Hi Everyone,
I am looking into AMI (using PHP) to record every instance of HOLD that is
generated by putting a caller on HOLD (press hold button on the phone set).
There is no HOLD in Asterisk but the event Music on Hold is generated when
HOLD is pressed. The complexity is that all of the the
From our experience it is not enough. We had to rewrite CDR generation to
suite our billing needs. That was on 1.4.xx, we are not using 1.6+
Regards,
Mindaugas Kezys
Kolmisoft UAB
VoIP Billing Solutions
e-mail: i...@kolmisoft.com
URL: http://www.kolmisoft.com
Find us on Facebook
-Original
On Thu, Nov 18, 2010 at 10:54:53PM +0100, Thorolf Godawa wrote:
since some time I am looking for a current and reliable solution to send
and receive faxes (probably fax-2-mail and mail-2-fax) in conjunction
with Asterisk.
[snip]
What are you using? mISDN, CAPI4linux, HylaFAX, IAXmodem,
I can't believe nobody uses cisco 7970 with asterisk to help with my issue.
2 sip lines registered:
Line 1: ext 260
Line 2: ext 160
How to get Line 2 blinking when Line 2 (ext 160) is called?
For some reason with my setup when I call Line 2 - Line 1 is blinking.
I use firmware 8.0.3
On Fri, Nov 19, 2010 at 12:04:47PM +0530, Aparna Narayan wrote:
I tried to perform call forward in asterisk by writing the following in the
dial plan.The data base is getting updated with the caller ID number how
ever the call is not getting forwarded.
[apps]
exten =
Post the germane portions of your xml. How does your phone register each
line button?
Cassius
From: Peter Kowalski kowalla...@gmail.com
Organization: GreatValueMart
Reply-To: kowalla...@gmail.com, Asterisk Users Mailing List -
Non-Commercial Discussion asterisk-users@lists.digium.com
Date:
No dice on finding a fix for this. I've been looking through the bug tracker
and through the config files and haven't found anything...
- Original Message -
From: Jonathan C. Bailey jbai...@co.marshall.ia.us
To: Asterisk Users Mailing List - Non-Commercial Discussion
Below is my xml button 1 and button 2 portion. Any help will be appreciated.
line button=1
featureID9/featureID
featureLabelPete(260)/featureLabel
proxyproxyip/proxy
port5060/port
name130/name
displayNamePeter/displayName
autoAnswer
autoAnswerEnabled2/autoAnswerEnabled
On Mon, Nov 22, 2010 at 11:24 AM, Peter Kowalski kowalla...@gmail.com wrote:
Below is my xml button 1 and button 2 portion. Any help will be appreciated.
line button=1
name130/name
authName130/authName
authPasswordpass/authPassword
contact7b452e87-4496-4762-e11f-b26751a1884b/contact
/line
I have done something similar; I am using SIP load 8.5.2. I use port 5060 on
both line buttons.
Cassius
From: Peter Kowalski kowalla...@gmail.com
Organization: GreatValueMart
Reply-To: kowalla...@gmail.com
Date: Mon, 22 Nov 2010 13:24:41 -0600
To: Cassius Smith cass...@cassius.org
Cc:
Solved!
Thank you Jonathan.
Like you suggested - I've changed port on both lines to 5060 and changed
contact so all: name, authName and contact are the same and it is working
like charm.
Thanks again,
Peter
-Original Message-
From: jthurma...@gmail.com [mailto:jthurma...@gmail.com] On
The Asterisk Development Team has announced the release of libpri 1.4.11.5.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/libpri/
The release of libpri 1.4.11.5 resolves several issues reported by the
community and would not have been possible
Hi,
I have a problem with dialing status.
I'm using Asterisk 1.6 and a patton 4554 gateway for ISDN calls.
When I call fixed telephone (not mobile phone) after few ringing the
status change to answer but the phone is still ringing, so if I hangup
before someone really answer, the call is
Thank you very much for help in finding the log.
I have the log now. I'd like to know what to look for in trying to figure out
how the
calls are getting originated. I'd be happy to shere all the information. I just
don't
want to post information on this public list that might show other people
Use IPTables to lock down your machine to only accept incoming connections from
your local network and from the particular IPs that you are expecting
connections from (such as your SIP trunk, maybe).
That is of course assuming that these calls are made by SIP.
Don't forget to also change all
On 11/22/2010 06:44 PM, Kevin Keane wrote:
Use IPTables to lock down your machine to only accept incoming
connections from your local network and from the particular IPs that
you are expecting connections from (such as your SIP trunk, maybe).
That is of course assuming that these calls are
UNSUBSCRIBE
On Tue, Nov 23, 2010 at 6:46 AM, antselva antse...@tiscali.it wrote:
Hi,
I have a problem with dialing status.
I'm using Asterisk 1.6 and a patton 4554 gateway for ISDN calls.
When I call fixed telephone (not mobile phone) after few ringing the
status change to answer but the
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